KR20110002467A - An electrical system for a speaker and its control - Google Patents

Abstract

An electrical apparatus is disclosed, comprising a frame, a speaker connected to the frame, a digital signal processor configured to communicate with the speaker to receive sound data and control data, and to control the speaker, wherein the digital signal processor is connected to the frame, And a lamp base coupler electrically connected to the speaker and the receiver, wherein the lamp base coupler is detachably connected to the power source when power is present. A method of generating a diffused sound field using a specially designed acoustic diffuser is provided. The diffused sound field adjusting method includes broadcasting at least one calibration sound signal through each of the plurality of speakers M in the acoustic system, and transmitting at least one calibration sound signal in a plurality of microphones located apart from each other at a listening position. Receiving and calculating each relative speaker placement angle relative to the listening position between each of the plurality of speakers in response to receiving at least one calibration acoustic signal in the plurality of microphones, wherein each of the plurality of speakers Their angular position is determined relative to the listening position and facilitates positioning of the virtual channel.

Description

An electrical system for a speaker and its control method

The present invention relates to a speaker, and more particularly to a speaker suitable for use in a hotel, restaurant, home or living space.

Music, sound and video recording media are rapidly becoming popular among consumers in general to be reproduced in homes and similar environments. Businesses such as restaurants and hotels also offer their customers music. Typically, speakers used in such systems are physically connected to a central amplifier and receive amplified analog voice signals transmitted therefrom. In some application systems, multichannel recording and playback is required, and the purpose is to generate surround sound using directional sound effects. To achieve this effect, different speakers must be able to receive different acoustic signals. Such reproduction of pre-recorded multichannel sounds is achieved by placing speakers at predetermined locations, where the listener experiences the full effect of such multichannel encoding at the given listening position. In addition, the sound coming from the speakers is required to be able to clearly identify the directional sound effect toward a predetermined listening position. Speakers are generally designed to emit sound from their front. Thus, obtaining an appropriate directional sound effect depends on the proper location of the speakers to direct the sound to a predetermined listening position. The installation of the entire system therefore requires separate wire connections from the central amplifier to the individual speakers, and careful placement of each of these speakers, resulting in satisfactory surround sound effects.

For example, proper playback in a typical living room of a movie encoded with Dolby 5.1 or DTS 5.1 sound (see FIG. 1 (prior art)) may result in front, center, and right speakers 102, 104, 106 being located in the listener position 108. Needing to be placed in a predetermined position with respect to the surround left and surround right speakers 110, 112 as well for the left and right of the listener position, respectively (where "channels" or "ideal" respectively). Channels ".

For channels driven by a central sound source, for example the receiver amplifier 114, a professional and aesthetic arrangement of the speakers is required, which leads to speaker cables leading from the central amplifier sound source to the respective speakers. It is necessary to place it in an interior space such as a wall space or a ceiling. Such speakers need to be carefully placed in consideration of two important points: the angle at which the speakers are placed with respect to the listening position and the direction in which the speakers are placed. The subwoofer position for such encoding, although not very important, requires connecting the speaker cable and / or the power cable. Assuming some users do not allow access to adjacent attic or underground spaces or do not provide hollow wall structures, the connection of such wires can be very difficult or expensive. For some consumers, such an installation may be impossible to achieve aesthetically. Even for speakers capable of receiving preamplified acoustic signals wirelessly, most speakers still need to be connected to a power source, typically requiring 120V to 230V of power, and have similar problems. Let's do it.

In other situations, such as a restaurant where only one sound track is heard on all speakers, connecting wires is cumbersome. Furthermore, because each speaker receives the same amplified analog sound signal, each speaker volume cannot be adjusted individually, providing the same noise level for all customers.

Therefore, it is easy to install the proper signal and power wires, so that proper sound broadcasting of popular encoding formats is possible, and there is no inconvenience or expensive disassembly and maintenance work according to the consumer's situation, and allows independent control of each speaker. There is still a need for an acoustic system.

An electrical apparatus is disclosed and includes a frame, a speaker connected to the frame, and a digital signal processor coupled to the frame, in communication with the speaker to receive sound data and control data, and to control the speaker. A lamp base coupler is electrically connected to the speaker and the receiver, and detachably connected to the power source, for example through a screwed base or an insertable and multi-pronged pin base. In the above embodiment, the speaker and the digital signal processors on the frame can be detachably connected to a power source through the lamp base coupler, so that the acoustic signal is individually controllable.

In one embodiment, the digital signal processor may receive acoustic data and control data using wireless radio frequency (RF) or power line communication technology.

In one embodiment, a method is provided for generating a diffused sound field through a specially designed acoustic diffuser.

In another embodiment, the electrical device is individually controllable in color of light, including a light electrically connected to the lamp base.

In another embodiment of the invention, the method of adjusting the sound field broadcasts at least one corrected sound signal through each of the plurality of speakers M in the acoustic system and at least within a plurality of microphones located away from each other at the listening position. Receiving one calibration acoustic signal and calculating each relative speaker placement angle relative to the listening position between each of the plurality of speakers in response to receiving at least one calibration acoustic signal in the plurality of microphones; The angular position of each of the plurality of speakers is determined relative to the listening position and facilitates positioning of the virtual channel.

 In an implementation of the invention, the method also receives a digital acoustic signal comprising a plurality of input digital acoustic signal channels (N) for generating an input acoustic channel amplitude vector representing one sound field, wherein each of the plurality of Determine an ideal virtual channel position for the listening position for input digital acoustic signal channels (N), and generate the one virtual output sound channel amplitude vector to simulate the ideal virtual channel position for the listening position. Rotate, and amplify the virtual output acoustic channel amplitude vector through the plurality of speakers M so that the plurality of input digital acoustic signals N are rotated for amplification through the plurality of speakers M to listen. Broadcasts are made within a sound system that simulates the ideal channel position for a location. .

The present invention allows easy installation of appropriate signal and power wires to enable proper sound broadcasting of popular encoding formats, and eliminates inconvenience or expensive disassembly and maintenance, depending on the consumer's situation, and enables independent control of each speaker. Provided are an electrical system for a speaker and a control method thereof.

The components presented in the figures are not necessary for accumulation, and are highlighted to illustrate the principles of the invention. Like numbers refer to like elements throughout the several views.
1 (Prior Art) is a block diagram of an acoustic system having five speakers arranged in an ideal channel position inside a room and broadcasting a 5.1 encoded acoustic signal to a listening position;
FIG. 2 is an exploded plan view showing one embodiment of a speaker and light assembly driven by the transmitter shown in FIG. 6;
3 is a plan view of the speaker and back assembly shown in FIG. 2;
4 is a block diagram of an embodiment, showing a receiver for receiving sound and control data from the transmitter shown in FIG. 6 to drive a speaker and control lighting;
5 is a block diagram of a sound system that, in one embodiment, enables the design of a sound output that includes multiple microphones to simulate an ideal channel placement for a listening position;
FIG. 6 is a block diagram of an embodiment, illustrating a transmitting device designed to design and transmit a multi-channel acoustic signal to adjust multiple acoustic channels and simulate an ideal channel arrangement for a listening position; FIG.
7 illustrates the generation of sound field design parameters that enable the simulation of an ideal channel arrangement for a listening position, as an example of a flow diagram;
8 illustrates a rotation matrix design for rotation of a multi-channel sound field, as an embodiment of a flow diagram;
FIG. 9 illustrates the use of the design parameters of FIGS. 7 and 8 to rotate the sound field to simulate ideal channel placement in an acoustic system with an ideal speaker arrangement, as an example of a flow diagram;
FIG. 10 is a block diagram of one embodiment, showing a sound system having the speaker and back assembly shown in FIGS. 2 and 3 and adapted to simulate a number of digital input sound channels to simulate an ideal channel placement for a listening position. ;
FIG. 11 is a block diagram of another embodiment, showing a sound system having the speaker and back assembly shown in FIGS. 2 and 3 and adapted to simulate a number of digital input sound channels to simulate an ideal channel placement for a listening position. .

2 illustrates one embodiment of a speaker and back assembly. The frame, preferably the speaker mounting bracket 202, houses the speaker 204 and the printed circuit board (PCB) 206 and is preferably located in the body housing 208 which provides thermal conduction of waste heat during operation. In one embodiment, receiver 400 (see below) is seated on PCB 206 and includes speaker electronics, such as a digital signal processor and amplifier (not shown), to drive speaker 204. Preferably, the body housing 208 is formed from a metal that facilitates thermal conduction to remove waste heat from the speaker electronics, such as aluminum. Preferably the two RF antennas 210 are connected to the PCB 206 on opposite sides of the speaker mounting bracket 202 to provide better signal diversity than that obtained from a single antenna. The upper and lower clamshell members 212 and 214 forming the acoustic diffuser 215 are connected to the speaker bracket 202 through a mounting bracket assembly 216. The acoustic diffuser is formed in a complementary opposite position with respect to the speaker 204 and creates a diffused sound field during operation. The lower shell-like member 214 is preferably of a conical or other shape in a predetermined shape to provide the desired sound diffusion.

In a preferred embodiment, an LED light 218 is seated in the diffuser assembly 215 to project light through the translucent decorative filter 220. The shell-like members 212 and 214 are preferably made of translucent gloss-removable polycarbonate or other thermoplastic polymers, glass or other resins, and are suitably translucent to light and are resistant to heat generated in proximity to the LCD or the like. Resist. The diffuser assembly 215 also preferably includes an aluminum coupler 222 between the upper and lower clamshell members 212 and 214 to provide thermal conduction of waste heat generated from the LED 218. The housing outer ring 224 is preferably formed as a translucent polyurethane material and seated about the near end 226 of the body housing 208 in the circumferential direction on the speaker bracket 202. The upper ring 228 is preferably formed as a translucent polycarbonate material and seated around the far end 230 of the body housing 208. In one embodiment, a lamp base coupler 232 is connected at the distal end 230 to the body housing 208 and detachably connected to a standard home or commercial power circuit. The lamp base coupler is preferably used in applications and national standards applicable to geographical areas of use, such as those used in Edison screw sockets ("E" bases), insertion holes, or 2- or 3-pin sockets. Appropriately available for the same multi-pronged pin base. Examples of 2- or 3-pin sockets are Type C (CEE 7/16, CEE 7/17), D (BS 546 5A / 250V), and M (BS 546 15A) used in India and other countries, and the United States. Type A (NEMA 1-15 USA 2 pin), and B (NEMA 5-15 USA 3 pin).

In one speaker and back assembly suitable for use in a home or restaurant environment, the various elements of the assembly as shown in FIG. 3 have an approximate size as shown in Table 1.

                Element               Length (mm)              D upper ring               40.5              D far end               35.3              H upper ring                14              H body housing               124.7              H outer ring                45              H assembly               194.6              H diffuser filter                70              H diffuser                59.5              D diffuser                52.4

4 shows a receiver 400 for use in the speaker and lamp assembly shown in FIGS. 2 and 3.

RF transmitter / receiver 402 and power line transmitter / receiver 404 are adapted to receive acoustic and control data from antenna 406 and receiver power line 408, respectively. Preferably, the RF transmitter / receiver 402 passes the processed digital sound signal through the processed digital sound signal path 409. The end user will control the data, for example volume, light or transmitter control data, which is received in the receiver controller 410 via the control data path 414 via the infrared receiver 412. In other embodiments, such end user control data may be received at the receiver 400 via the RF transmitter / receiver 402.

The back controller 416 communicates to the receiver controller 410 via the back control data path 418 and controls the lighting within the speaker and back assembly 200, for example LEDs 718 (see FIG. 7). do. Receiver acoustic amplifier 420 is coupled to digital signal processor 1006 via digital acoustic signal path 422 to receive digital acoustic signals for amplification to speaker 204 (not shown). The receiver acoustic amplifier 420 also communicates to a receiver controller 410 and directs the DSP control data path 411 by control data via the receiver controller data path 424, for example by the receiver controller 410. Receive increase / decrease volume control data received from the digital signal processor 406 via or from the end user via the infrared receiver 412. In one embodiment, back control data may be received from the digital signal processor 406 via the receiver controller 410 and provide an image associated with the sound signal correlated to volume or frequency characteristics of the digital sound signal. do.

FIG. 5 illustrates the use of multiple microphones 502 within the room shown in FIG. 1 for the first time, using a multi-channel sound field that simulates ideal channels using speakers placed at positions away from a predetermined ideal channel position. It allows the design of acoustic parameters for rotation. The ideal left, center and right channels 102, 104, 106 and the ideal surround left and right channels 110, 112 are shown in dashed lines, respectively, showing the ideal position relative to the listener position 108. To facilitate the description of one embodiment of the algorithm, it will be described that arbitrary speaker placement positions are shown as solid lines, and that 5.1 channel surround sound encoding signals are used. For example, the front left and front right speakers 504, 506 are shown in a position away from the ideal center channel 104 farther than predetermined for ideal channel placement of 5.1 surround sound. Similarly, surround left and surround right speakers 508 and 510 are shown in solid lines, and are shown at locations off-site for reproduction of the 5.1 channel surround sound encoded signal. A sound source 512 is arranged to communicate with the speakers 504, 104, 506, 508, 510 to communicate analog sound and data signals through a physical connection, eg, a home power wiring system. Or preferably, acoustic and data signals are sent to the respective speakers, which transmit such acoustic and control signals using an RF wireless transmitter and receiver (not shown) in the sound source 512. . Also shown are a plurality of microphones 502, each of which is spaced apart from one another, disposed around a listening position, and in communication with the sound source 512 via a microphone cable 513 to initially design acoustic parameters. It is possible to rotate the multi-channel sound field to simulate the ideal channel arrangement, as will be explained later.

5 and 6, the sound source 512 is, in one embodiment, a transmitter 600 having analog / digital converters (“A / D converters”) to receive analog sound data 604. For example, it can be received from an RC connector, an audio jack or a mini-DIN connector, and converts analog sound signals into digital sound signals. A digital acoustic receiver 606 is also preferably provided within the transmitter 600 to receive a digital acoustic signal 608, for example a digital coaxial acoustic connector, a toslink connector, an IEEE 1394 interface, or other suitable Receives from a digital acoustic connection device a standard, de facto standard or proprietary digital acoustic and control data signal. Digital acoustic signal paths 610 and 612 are provided to the A / D converter 602 and the digital acoustic receiver 606 respectively to communicate digital acoustic signals to the digital signal processor 614. The digital signal processor 614 consequently transmits the processed digital sound signal to the processed digital sound signal path 616 to the air via a radio frequency (RF) transmitter / receiver 618 or to a power line transmitter. / Receiver 620 to be transmitted over the power line. The processed digital sound signal may also be converted to an analog sound signal 622 using a digital / analog converter 624 and provided to an analog output terminal (not shown). Control data paths 626 and 628 respectively connected to the A / D converter 602 and the digital acoustic receiver 606 enable communication of control data to the transmitter controller 630 side.

During operation, the transmitter controller 630 preferably sends control data information to the digital signal processor 614 for proper processing of the digital acoustic signal, which is from the A / D converter 602 and the digital acoustic receiver 606. Enter the digital signal processor 614. For example, the digital acoustic receiver 606 may communicate information with the transmitter controller 630, and provide a signal encoding method, such as a PCM or Dolby encoding method, to control data from the digital acoustic receiver 606. Appropriate sampling of the digital acoustic signal provided to the digital signal processor 614 side via the path 612. The A / D converter 602 provides sampling rate information for the transmitter controller 630 via a control data path 626 to provide appropriate control data to the digital signal processor 614 and the A The digital sound signal is received from the / D converter 602.

A microphone amplifier 632 communicates to the A / D converter 602 via an analog acoustic data path 636 and moves the microphone signal 634 to the digital signal processor 614 for the design of acoustic parameters. In one embodiment of the invention, the rotation of the multi-channel sound field is allowed.

In one embodiment of the present invention, an R / F radio transmitter / receiver 618 is included, and an antenna 638 is connected to the R / F radio transmitter / receiver 618 via an R / F signal path 640. And receive R / F signals with sound and control data. The RF receiver or preferably the infrared (IR) receiver 642 is intended for transmitter 600 control data, such as volume, sound source selection, surround acoustic encoding selection, lighting control (for additional placement) or other receivers. An infrared signal 644 containing end user information is received and communicated to the transmitter controller 600 via the control data path 646.

In one embodiment of the operation shown in FIG. 7, the transmitter 600 performs calculation of design parameters to allow rotation of the multi-channel sound field and simulate ideal channel placement. In anticipation of a non-ideal multiple speaker arrangement as shown in FIG. 5, the digital signal processor initializes the speaker coefficients to a value of 1 (block 700). If the loudspeaker coefficient is not equal to 1 plus the number of speakers previously detected by the digital signal processor (block 702), one or more acoustic signals are a subject speaker ("calibrated acoustic signal"). Is preferably broadcast on an acoustic signal frequency sweep (block 704). The broadcast calibration acoustic signal is received via a plurality of microphones positioned at a listening position (block 706) and provided to a digital signal processor. In a preferred embodiment, three microphones are placed at the corner points of an equilateral triangle approximately 6 cm apart on one plane to sense the physical placement of the target speaker by the digital signal processor within two dimensions. Alternatively, four microphones are used in tetrahedrons spaced at equal distances, for example approximately 6 cm apart, to sense the target speaker in three dimensions. An impact response for the broadcast calibration acoustic signal is calculated, preferably by taking an FFT of the frequency sweep signal and an inverse Fourier transform (FFT) of the FFT ratio of the received microphone signal (block 708). . A crossover ("Xover") filter is calculated, which is the fourth rank Butterworth filter, whose cutoff frequency is calculated from the previously calculated frequency response of the impact response (block 710). Preferably, the point at which the amplitude of the frequency bombardment is lowered to -10Db of the maximum amplitude over the entire frequency range is obtained as the cutoff frequency. The fourth rank low pass coefficient and the fourth rank Butterworth high pass filter coefficients are calculated. Using the plurality of microphones described above, the target speaker angle and height are calculated relative to the listener position (microphone position) (block 712). More specifically, using the shock response of each microphone, the time difference [Delta] t between the peak amplitudes of the shock responses between all microphone pairs is first calculated. The time difference Δt is utilized to give an angle of incidence in the acoustic direction. For example, a zero second time difference Δt indicates that the sound has reached two paired target microphones simultaneously, so that the source is a hyper-plane at the same distance from the two microphones. It is located at. Similarly, a time difference [Delta] t equal to the time obtained acoustically to cover the distance between the two microphones indicates that the sound source is in a straight line connecting the two target microphones. The incoming acoustic angle in relation to the straight line connecting the two microphones is calculated as an inverse cosign of the ratio Δt over time obtained as the sound to cross the distance between the two target microphones. Each such angle represents a possible hyper-plane, where a target speaker broadcasting the calibration signal can be located relative to the target pairs of the speakers. The physical location of the target speaker relative to the listening position is placed using data obtained from a number of such microphone pairs. The physical location giving the least error for all the computed hyperplanes is obtained as the location of the broadcast speaker. Using Cartesian coordinates of the broadcast sound source, the target speaker angle in the horizontal plane is calculated in relation to the front and the height.

In response to receiving a calibration signal broadcast through the target speaker, the loudness of the target speaker is calculated to calculate level compensation (block 714), which is accomplished by calculating the average of all frequency response magnitudes for the target speaker. . The reverse of this process is utilized to match the volume of each subsequent speaker. Delay compensation is calculated (block 716), which is first done by calculating a delay that occurs between the calibration signal broadcast and the reception of such a signal at the microphone, preferably the impact rejection being maximized. This is done by inspection of the point. This delay is then subtracted from the predetermined maximum delay allowed by the system and used as a delay compensation factor. An EQ filter is calculated for the target speaker (block 718), for later compensation of all irregular frequency responses of the predetermined shock response. The shock response is first passed through a set of all pass filters to mimic the nonlinear frequency scale of the human auditory system. Then, the magnitude (m) of this modified impact response is calculated using the FFT. The finite impact response (FIR) (iw) is calculated, which is the minimum phase filter, whose magnitude response is the opposite of m. The FIR (iw) is then passed through a set of all pass filters, inverting the nonlinear mapping to yield the final EQ filter.

The speaker coefficients are increased (block 720) and the speaker coefficients are compared again with the maximum speakers in the acoustic system. If the speaker coefficient does not match the Max + 1 speakers, the process is preferably repeated, broadcasting one or more calibration acoustic signals through the next target speaker (blocks 702, 704). Or if the speaker coefficient matches Max + 1 speakers (block 702), then the next step of the design process is to calculate the rotation matrix as a digital signal processor (block 722), as described above, at block 712 Use the generated speaker angle and height data.

In FIG. 8, a flow diagram illustrates one embodiment of a rotation matrix design for rotation of a multi-channel sound field. The number of input digital acoustic signal channels is determined (block 802) to determine the relative position of the ideal virtual channels relative to the listening position (block 804). For example, a Dolby 5.1 or DTS 5.1 system is located on both sides of the center channel and is defined by left and right front speakers 1.5 m away. The left and right surround speakers are arranged at positions which are also approximately 1.5m apart on both sides of the listening position. In response to the capture of broadcast calibration acoustic signals at all microphones, the speaker pairs s1, s2 closest to both sides of the ideal virtual target channel position are calculated from the calculated speaker angles (block 806). If the system successfully calculates the closest pair of speakers (block 808), then the angle difference between the speakers s1 and s2 and the ideal virtual target channel positions are determined (la1, la2, respectively). (Block 810) (see FIG. 5). For example, as shown in FIG. 5, the front left speaker 504 and the center speaker 104 represent the speakers s1 and s2, respectively. The angles la1 and la2 represent the angle difference between the speakers s1 and s2 and the ideal virtual target channel position, respectively, approximately 16.6 degrees and 39.7 degrees, respectively. In another embodiment for a sound system capable of determining the speaker position in three dimensions, the 3-D angular difference la1, la2 between the speakers s1 and s2 and their respective ideal virtual channel positions are determined (block 812). ). Speaker coefficients g1 and g2 are calculated for the 2-D relationship for speakers s1 and s2 (block 814):

(1) sqrt (gl * gl + g2 * g2) = l

(2) g1 / g2 = cos (lal) / cos (la2)

Next, the M x N rotation matrix is located with speaker coefficients (block 816).

If the acoustic system cannot calculate the closest pair of speakers s1 and s2 according to the above description (block 808), then the column N for the ideal target channel of the M × N rotation matrix is 1 / sqrt (M). The digital sound input amplitude is evenly distributed across the target speaker with a coefficient set to (block 818).

In one embodiment using the rotation matrix shown in FIG. 8, FIG. 9 is an embodiment of a flow diagram, in which the design is adapted to rotate the sound field for simulating an ideal channel arrangement in an acoustic system with an ideal speaker arrangement. Illustrates the use of parameters. The input digital acoustic signal channels N of the digital acoustic sample 900 are passed through respective crossover filters 902 to form an input acoustic channel amplitude vector 904, which is shown in the flow diagram of FIG. 8. As described, multiplied by rotation matrix 906 produces a virtual output speaker channel amplitude vector 908. Speaker channels 1 to M are introduced in the 2-D embodiment of the rotation matrix 906, preferably via additional acoustic compensation filters, for example each delay compensation block 910, level compensation block 912. And the resulting digital sound signals 1 through M 916 introduced into the EQ filters 914 are amplified and broadcasted through respective speaker channels.

In another embodiment configured for a 3-D rotation matrix (not shown), the delay compensation blocks may be omitted, as a result of the three-dimensional and angular difference calculations available for each speaker channel 1 to M without additional delay compensation. have.

FIG. 10 illustrates an embodiment of a multi-channel acoustic system, which uses the speaker and back assembly shown in FIGS. 2 and 3 and is not ideal for adjusting sound fields with multiple digital input acoustic channels. Simulate an ideal channel layout using speakers placed on the Front left, front right, center, left surround and right surround speaker and lighting assemblies 1002,1004,1006,1008,1010, respectively, are detachably attached to the speaker and lighting assemblies, as shown in FIGS. It is shown as a connected standalone light stand. Such loudspeakers and lighting assemblies are suitable for applications applicable to geographic areas of use and for lamps suitable for national standards, for example Edison screw sockets ("E" bases) or inserts ("B" bases). It can be seen that a coupler can be used.

FIG. 11 illustrates another embodiment of a sound system installed in a room, using the speaker and lamp assembly shown in FIGS. 2 and 3. Speaker and back assemblies 200 are shown as detachably coupled to freestanding light poles for left front, center and right front speakers 1002, 1004, 1006.

In such an embodiment, the speaker and back assembly 200 is fixed to the left surround wall protruding stand 1102 and the right surround wall protruding stand 1104, and preferably receives an RF signal. Alternatively, the speaker and back assemblies 200 may receive sound and control data from a power line in a room electrically connected to the sound source 512.

While various embodiments and implementations of the invention have been described above, it will be apparent to those skilled in the art that many more embodiments and implementations can be made within the scope of the invention.

Claims (21)

frame;
A speaker connected to the frame;
A digital signal processor in communication with the speaker, receiving sound data and control data, controlling the speaker, and connected to the frame; And
And a lamp base coupler electrically connected to the speaker and the receiver, and when a power source is present, detachably connected to the power source.
And said speaker and digital signal processors on said frame can be detachably connected to a power source through said lamp base coupler. 2. The electrical device of claim 1 wherein the lamp base coupler is of a type selected from the group such as screwed bases, insert-mounted and multi-pronged pin bases. 3. The electrical device of claim 1, further comprising a light electrically connected to the lamp base coupler. 4. 3. The electrical device of claim 1 or 2, further comprising an antenna in communication with the digital signal processor to receive radio frequency (RF) acoustic data and control data. 5. The apparatus of claim 4, further comprising a transceiver coupled between the antenna and the digital signal processor to receive radio frequency (RF) control and sound data and to transmit radio frequency (RF) control data. Electrical device. 3. The electrical device of claim 1 or 2, further comprising a power line transceiver coupled to the digital signal processor for transmitting control and acoustic data. The apparatus of claim 1, further comprising a body housing that provides thermal conduction of waste heat, wherein the body housing surrounds the digital signal processor and an amplifier for driving the speaker. Electrical devices. 8. An electrical device as claimed in any preceding claim, further comprising an infrared receiver which receives control data of an end user and transmits it via the transceiver coupled to the digital signal processor. 9. The electrical device of claim 8 further comprising a back controller in communication with the light, the back controller being operable to control light output of the back. 10. An electrical device as claimed in any preceding claim, further comprising an acoustic diffuser disposed in a complementary position with respect to the speaker and generating a diffused sound field during operation of the speaker. 11. The electrical device of claim 10 wherein the acoustic diffuser comprises a conical acoustic diffuser. 11. The electrical device of claim 10 wherein the lamp is located inside the acoustic diffuser. The electric device according to any one of claims 1 to 12, wherein the lamp is an LED lamp. 18. The electrical device of claim 13 wherein the LED light is operable to selectively change color. The electrical device of claim 13, wherein the LED light is operable to selectively change color in response to amplitude or frequency characteristics of music. Broadcast at least one calibration acoustic signal through each of the plurality of speakers M in the acoustic system;
Receive the at least one calibration acoustic signal in a plurality of microphones located remotely from each other in a listening position; And
Calculating each relative speaker placement angle relative to the listening position between each of the plurality of speakers in response to receiving the at least one calibration acoustic signal in the plurality of microphones,
Wherein the angular position of each of said plurality of speakers is determined relative to said listening position and facilitates positioning of the virtual channel. 17. The method of claim 16, further comprising: receiving a digital acoustic sample comprising a plurality of input digital acoustic signal channels (N) to produce an input acoustic channel amplitude vector representing one sound field;
Determine an ideal virtual channel position for the listening position for each of the plurality of input digital acoustic signal channels (N);
Rotate the sound field to produce one virtual output acoustic channel amplitude vector to simulate the ideal virtual channel position relative to the listening position; And
Amplifying the virtual output acoustic channel amplitude vector through the plurality of speakers (M),
And the plurality of input digital sound signals (N) are rotated for amplification through the plurality of speakers (M) to broadcast in an acoustic system that simulates an ideal channel position for the listening position. . The method of claim 17, wherein rotating the sound field,
Mapping the input digital acoustic signal channels (N) to the plurality of speakers; And
And multiplying the input acoustic channel amplitude vector by the mapping to generate the virtual output speaker channel amplitude vector. The method of claim 18, wherein the mapping,
For each ideal virtual channel position, calculate the closest pair of speakers (s1, s2) selected from the group of the plurality of speakers (M) on opposite sides of the ideal virtual channel position;
For the listening position, calculate a respective angular difference (la1, la2) between the respective speakers (s1, s2) and the respective ideal virtual channel position;
Calculate the rotation matrix coefficients g1, g2 according to the following correlation;
sqrt (gl * gl + g2 * g2) = l
g1 / g2 = cos (lal) / cos (la2)
Positioning an M x N rotation matrix having at least the coefficients g1 and g2 in cells M1N1 and M2N1, respectively. 18. The sound field adjusting method of claim 17, wherein the broadcasting is performed continuously through each of the plurality of speakers. 18. The sound field adjustment method of claim 17, wherein the at least one calibration acoustic signal comprises a frequency sweep.

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